1. Field of the Invention
The present invention relates to video and audio signal multiplex sending apparatus and receiving apparatus and a transmitting apparatus formed by combining them for sending and receiving video and audio multiplex signals for multiplexing an audio signal in a blanking period of a video signal.
2. Related Art of the Invention
In transmission and recording of video signals or audio signals, conventional analog systems have been more often replaced by digital systems smaller in image quality deterioration and sound quality deterioration. Specifically for a system used in a studio, a standard of STMPTE259M is provided for a system of transmitting video and audio signals through a transmission path (for example, "2-1-3. Digitization of Transmission in a Station, (1) 10B Scramble System"). It is a standard for serial digital interfacing, and a audio signal is multiplexed in a blanking period of a video signal, and transmitted according to the STMPTE259M system.
By referring to the drawings, a conventional video and audio signal multiplex transmitting apparatus of the SMPTE259M system is described below.
FIG. 11 is a block diagram showing a constitution of a conventional video and audio signal multiplex sending apparatus. The sending apparatus comprises a buffer memory 5 for temporarily storing an audio signal, an audio clock oscillator 3 for outputting an audio clock, a video clock oscillator 4 for outputting a video clock and a signal multiplexer 6 for multiplexing the video and audio signals.
In FIG. 11, a digital video signal is applied to the signal multiplexer 6 through a video signal input terminal 1. On the other hand, a digital audio signal is applied through an audio signal input terminal 2, temporarily stored in the buffer memory 5 according to an audio clock outputted by the audio clock oscillator 3, then, read out of the buffer memory 5 according to a video clock outputted by the video clock oscillator 4, and the audio signal read is applied to the signal multiplexer 6. In the operation, because the frequency of video clock is higher than that of the audio clock, thus, reading of the audio signal is faster than writing, the reading of the audio signal is stopped temporarily, and the audio signal data is multiplexed only in a blanking period.
The signal multiplexer 6, after multiplexing the audio signal read out of the buffer memory 5 in a blanking period of the video signal that is applied through the video signal input terminal 1, outputs the multiplex signal to a multiplex signal output terminal 7. Here, the video clock outputted from the video clock oscillator 4 is synchronous with the frequency of video signal, and the audio clock outputted from the audio clock oscillator 3 is synchronous with the audio signal input.
The buffer memory 5 is employed between the audio signal input terminal 2 and the signal multiplexer 6, because the video and audio signals are digitized at different frequencies, and it is required to convert the frequency of audio signal to that of the video signal, which is achieved by storing the audio signal in the buffer memory 5 by means of the audio clock, and read out of the buffer memory 5 by the video clock.
FIG. 12 is a block diagram showing a constitution of a conventional video and audio signal multiplex receiving apparatus. The receiving apparatus comprises a signal separator 12 for separating video and audio signals of a multiplex signal, a buffer memory 14 for temporarily storing the audio signal separated, a video clock oscillator 15 for outputting a video clock, a write address generator 16 for generating a write address according to the video clock, an audio clock oscillator 17 for generating an audio clock, a read address generator 18 for generating a read address according to the audio clock, a phase comparator 19 for comparing the write and read addresses and the like.
In FIG. 12, a multiplex signal applied through a multiplex signal input terminal 11 (for example, a multiplex signal outputted from a video and audio signal multiplex sending apparatus of FIG. 11) is separated to video and audio signals by the signal separator 12, and the video signal is outputted to the video signal output terminal 13. The other signal, that is, the audio signal is required to be converted to a frequency identical with that of an original audio signal, because it has been converted to a frequency same as that of the video signal in the sending side. Thus, the audio signal separated is temporarily stored in the buffer memory 14 according to a write address generated by the write address generator 16 by means of a video clock outputted from the video clock oscillator 15. In the operation, only the audio signal is selected and written by suspending the writing operation, in contrast with the case of sending a signal. The audio signal stored in the buffer memory 14 is read according to a read address outputted by the read address generator 18 by means of an audio clock that is outputted by the audio clock oscillator 17. In order to recover an audio clock at a frequency identical with that of the signal sent, the phases of write and read addresses are compared with each other by the phase comparator 19, and a clock generated by the audio clock oscillator 17 is controlled according to a result of the comparison. Here, a circuit formed by the audio clock oscillator 17 and the phase comparator 19 provides a phase locked loop (hereinafter referred to as PLL). An audio signal read out of the buffer memory 14 is outputted to an audio signal output terminal 20.
In such multiplex transmission system as described above, however, it is a problem that the circuit is increased in size, since the recovery of a clock is affected by an accumulation of the buffer memory, in order to control the effect, and a phase relation of an audio signal with that of a video signal cannot be stored in the sending and receiving sides, as a residual phase change is caused in a clock smoothed by the PLL circuit, because a voltage level controlled by the PLL can be adjusted only by a phase difference between the R and W addresses of buffer memory, and a fine adjustment is unavailable. This is a first problem that the invention is to solve.
S17.100 of SMPTE also provides a standard for multiplexing audio digital data or an additional data in a supplemental data area for serial digital video signals according to a standard of SMPTE 259M. In other words, the standard provides for transmitting a digital audio signal in a blanking period of a video signal which is a main signal. As for an audio system, provisions of AES3-1991 (ANSI S4.40-1991) are applied correspondingly. According to the standard for transmitting a digital audio signal as a serial digital signal, three packets are provided: control packet, audio packet and additional packet.
FIG. 13 shows data formats of the three packets. FIG. 13(a) shows a data format of the control packet. Thus, the control packet comprises a sequence of words of 10 bits, and is fixed to seventeen words for a composite system and nineteen words for a component system. An ANC data flag (ADF) is of one word (3FCh) for a composite system and three words (000h, 3FFh, 3FFh) for a component system. Data ID (DID) indicates to which of audio groups 1, 2, 3 and 4 the control packet pertains. A data block number (DBN) is constantly fixed to 200h. Data count (DC) is constantly fixed to 20Ch (twelve words). ATF1-2 shows an audio frame number for ch1 and ch2. AFT3-4 shows an audio frame number for ch3 and ch4. RATE specifies a sampling frequency of the audio ch pairs. ACT shows an active ch. DELA0 to DELA2 or DELB0 to DELB2 shows a relative delay of an audio signal to a video signal expressed by a multiple of sampling interval. CS is an error detection code of the control packet.
The control packet is always transmitted once before each field. By means of the control packet, a sampling frequency, synchronization or non-synchronization with a video signal, delay in relation with the video signal, presence or absence of an audio signal and audio frame number are provided. If the control packet is not received, it is determined that the audio data is in synchronization with the video signal at a sampling frequency of 48 kHz. The sampling frequency is set to either 32, 44.1 or 48 kHz according to a transmission speed of the audio packet.
FIG. 13(b) shows a data format of the audio packet. Thus, the audio packet comprises a sequence of words of 10 bits, and the number of words is variable. An ANC data flag (ADF) is one word (3FCh) for a composite system and three words (000h, 3FFh, 3FFh) for a component system. Data ID (DID) shows to which of audio groups 1, 2, 3 and 4 the audio packet pertains. Data block number (DBN) is a serial number applied to audio packets that pertain to a same audio group, when audio signals sequentially applied are grouped to several audio packets, and has a value periodically changed in a range of 1 to 255 assigned thereto. Data count (DC) indicates the number of words contained in user data. The user data is a subframe AD1 or AD2 comprising units of three words, and the number of words of the user data is 255 at the maximum. The subframe AD1 or AD2 has data concerning digital audio data of 20 bits in the MSB side assigned thereto. To CS, an error detection code of the audio packet is assigned.
Now, contents of bit addresses of 30 bits of the subframe AD1 or AD2 are shown in FIG. 14(a). Thus, a bit sync (Z) is for showing whether the subframe is followed by a new channel status block. If it is followed by a channel status block, then, Z=1, and if not, then, Z=0. Ch1 and ch2 are for identification of audio ch1 to 4. Moreover, aud0 to aud19 are digital audio data of 20 bits expressed linearly by a complement of 2. A validity bit (V) indicates the validity of an audio sample, and V=1, if digital audio data of a subframe is suitable for conversion to an analog audio signal, while V=0, if it is not. A user bit (U) is for transmitting user data specified by a user. A channel status bit (C) is for transmitting information related to an audio channel, and a block consists of channel status bits corresponding to a 192 bits. As described above, the block sync of a subframe followed by the block is at Z=1. A parity bit (P) is of an even number for 26 bits of a subframe, excluding those of b9 in the first, second and third columns.
FIG. 13(c) shows a data format of the additional packet. Thus, the additional packet comprises a sequence of words of 10 bits, and the number of words is variable. An ANC data flag (ADF) is one word (3FCh) for a composite system and three words (000h, 3FFh, 3FFh) for a component system. Data ID (DID) shows to which of audio groups 1, 2, 3 and 4 the additional packet pertains. Data block number (DBN) is a serial number applied to additional packets that pertain to a same audio group, when 4 bits in the LSB side of digital audio data of an audio group are subgrouped, and has a value periodically changed in a range of 1 to 255 assigned thereto. Data count (DC) indicates the number of words in AUX column. Data concerning to digital audio data of 4 bits in the LSB side is assigned to the AUX consisting of words of 10 bits. CS is an error detection code of the additional packet.
Now, contents of bit addresses of 10 bits of the AUX are shown in FIG. 13(b). Thus, 4 bits in the LSB side of the subframe AD1 are assigned to x0 to x3. The LSB of them is assigned to x0. Then, 4 bits in the LSB side of the subframe AD2 are assigned to y0 to y3. The LSB of them is assigned to y0. P is an even number parity for b0 to b7.
An apparatus completely meeting the standard is operated in two modes: 24-bit mode for transmitting 24-bit audio signals and 20-bit mode for transmitting 20-bit audio signals. In the 24-bit mode, data corresponding to 20 bits in the MSB side thereof is transmitted by the audio packet, and the balance corresponding to 4 bits by the additional packet. Then, in order to simplify a circuit for reproduction in the AES format, audio and additional packets must be transmitted in a same blanking period. Besides, the audio and additional packets are required to be transmitted adjacently. In this mode, audio signals of an accuracy of 24 bits can be transmitted in four to twelve channels. In contrast, in the 20-bit mode, all 20 bits can be transmitted by the audio packet, the additional packet is unused. In this mode, audio signals of an accuracy of 20 bits can be transmitted in four to sixteen channels.
According to such conventional method, however, if the audio packet is lost due to an error in a transmission path, interpolation of the lost packet is unachievable, because the length of the packet, which is variable, is unpredictable. It has been, therefore, a problem that such information as of audio signals may be caused, leading to a loud noise. This is a second problem that the invention is to solve.